Speech transmitter

ABSTRACT

A voice transmission apparatus  100  includes a subscriber-side I/F processing unit  1 , a voice signal processing unit  2 , a packet processing unit  3 , an Ethernet I/F processing unit  4 , a device-operation-monitoring-control processing unit  5 , an amount of discarded voice packets/packet jitter measuring unit  6 , a calling restriction determining unit  7 , and an ITU-T Q.50 DLC processing unit  8 . An operation monitor control desk  9  and an exchange  10  are connected to the voice transmission apparatus  100 . The voice transmission apparatus  100  can impose a calling restriction on the exchange  10  based on the amount of discarded packets and the amount of packet jitter measured by the amount of discarded voice packets/packet jitter measuring unit  6.

FIELD OF THE INVENTION

The present invention relates to a voice transmission apparatus whichcan be placed between a subscriber-side exchange to whichsubscriber-side apparatus are connected and a network.

BACKGROUND OF THE INVENTION

A voice transmission apparatus which is an example of prior artmultiplexing transmission apparatus has a structure in which a pluralityof 2.048 MHz (E1) or 1.544 MHz (T1) signal lines according to ITU-T(International Telecommunication Union-Telecommunication Sector)recommendation G.703/G.704 from a subscriber-side exchange are inputtedto a subscriber-side interface processing unit that interfaces with thesubscriber-side exchange. 30 channels are multiplexed per one voicesignal when the plurality of signal lines inputted to thesubscriber-side interface processing unit are E1 signal lines, whereas24 channels are multiplexed per one voice signal when the plurality ofsignal lines inputted to the subscriber-side interface processing unitare T1 signal lines. The prior art voice transmission apparatusdisassembles incoming voice signals channel by channel, and detectssound portions of each of the incoming voice signals, encodes andcompresses the sound portions of each of the voice signals, assemblespackets from them, and sends out them onto a network, e.g., an IP(Internet Protocol) network.

As mentioned above, the prior art voice transmission apparatus has afunction of converting voice signals transmitted thereto via E1/T1channels into IP packets, and is so constructed as to send out all soundportions of the voice signals transmitted thereto via the E1/T1channels, as IP packets, onto a network. However, in such a prior artvoice transmission apparatus, a large volume of data exists in an IPnetwork and network congestions can occur. In order to carry outhigh-quality data transmission even in this case, various techniqueshave been proposed. For example, Japanese patent application publicationNo. 8-251,226, Japanese patent application publication No. 7-303,114,and Japanese patent application publication No. 4-3,544 disclose priorart techniques for carrying out high-quality data transmission even whena large volume of data exists in an IP network and network congestionscan occur.

In such a prior art voice transmission apparatus, since all soundportions of voice signals inputted via E1/T1 channels are outputted, asIP packets, to a network as mentioned above, and data packets from otherdata terminals are further inputted into the network, the amount ofpackets transmitted via the network can increase extremely. In such acase, routers and so on arranged in the network may discard voicepackets, or a fluctuation, which is so-called packet jitter, may occurwithin a transmission time period during which voice packets aretransmitted. A voice transmission apparatus which has received such asequence of packets cannot send out a voice signal which is normallydecoded to a subscriber-side exchange and this causes degradation in thequality of voice signals.

DISCLOSURE OF THE INVENTION

In accordance with the present invention, there is provided a voicetransmission apparatus arranged between a subscriber-side exchange towhich a subscriber-side terminal apparatus is connected and a network,the voice transmission apparatus including: a first interface means forexchanging a signal with the subscriber-side exchange; a secondinterface means for exchanging a signal with the network; a voice signalprocessing means for detecting sound portions of a voice signal whichconstitutes the signal, for encoding the voice signal having soundportions, and for decoding an encoded voice signal inputted thereto; apacket processing means for assembling packets from the voice signalencoded by the voice signal processing means, and for disassemblingpackets inputted thereto into a signal and for supplying it to the voicesignal processing means; a measurement means for measuring an amount ofpackets which are discarded from packets associated with a voice signalincluded in the received packets, and an amount of jitter of the packetsassociated with the voice signal; a calling restriction determinationmeans for determining whether to impose a calling restriction based onmeasurement results obtained by the measurement means; and a controlsignal creating means for creating a control signal for imposing acalling restriction on the subscriber-side exchange when the callingrestriction determination means determines that the calling restrictionneeds to be imposed on the subscriber-side exchange.

Therefore, the present invention offers an advantage of being able toprevent degradation in the quality of voice signals by imposing acalling restriction on the subscriber-side exchange when networkcongestions occur due to a large amount of data packets and this resultsin a large amount of discarded voice packets and a large amount ofpacket jitter.

In the voice transmission apparatus according to the present invention,the packets include an IP packet, a UDP packet, and an RTP packet.

Therefore, the present invention offers another advantage of being ableto prevent degradation in the quality of voice signals in the voicetransmission apparatus.

In the voice transmission apparatus according to the present invention,the second interface means is an interface means which complies withEthernet.

Therefore, the present invention offers a further advantage of beingable to connect the voice transmission apparatus with Ethernet.

In the voice transmission apparatus according to the present invention,the control signal output means is an ITU-T Q.50 DLC processing meansfor outputting the control signal for imposing a calling restriction onthe subscriber-side exchange according to ITU-T recommendation Q.50.

Therefore, the present invention offers another advantage of being ableto prevent degradation in the quality of voice signals in the voicetransmission apparatus.

BRIEF DESCRIPTION OF THE FIGURES

FIG. 1 is a block diagram showing the structure of a voice transmissionapparatus in accordance with embodiment 1 of the present invention;

FIG. 2 is a diagram showing the structure of the header of an RTPpacket;

FIG. 3 is a diagram showing the structure of a first communicationsystem in which one voice transmission apparatus in accordance withembodiment 1 of the present invention is arranged;

FIG. 4 is a diagram showing the structure of a second communicationsystem in which one voice transmission apparatus in accordance withembodiment 1 of the present invention is arranged; and

FIG. 5 is a flow chart showing a flow of the operation of one voicetransmission apparatus in accordance with embodiment 1 of the presentinvention.

PREFERRED EMBODIMENT OF THE INVENTION

In order to explain the invention in greater detail, the preferredembodiment of the invention will be explained below with reference tothe accompanying figures.

Embodiment 1

FIG. 1 is a block diagram showing the structure of a voice transmissionapparatus 100 in accordance with embodiment 1, and FIG. 2 is a diagramshowing the structure of the header of an RTP packet. FIG. 3 is adiagram showing the structure of a first communication system in whichone voice transmission apparatus 100 according to this embodiment isdisposed, and FIG. 4 is a diagram showing the structure of a secondcommunication system in which one voice transmission apparatus 100according to this embodiment is disposed. FIG. 5 is a flow chart showinga flow of the operation of one voice transmission apparatus 100according to this embodiment.

As shown in FIG. 1, the voice transmission apparatus 100 in accordancewith embodiment 1 is provided with a subscriber-side I/F processing unit1, a voice signal processing unit 2, a packet processing unit 3, anEthernet I/F processing unit 4, a device-operation-monitoring-controlprocessing unit 5, an amount of discarded voice packets/packet jittermeasuring unit 6, a calling-restriction determining unit 7, and an ITU-TQ.50 DLC processing unit 8, and an operation monitor control desk 9 andan exchange 10 are connected to the voice transmission apparatus 100.

In order to exchange a signal between the exchange 10 and the voicetransmission apparatus 100, the subscriber-side I/F processing unit 1(referred to as the PRI (Primary Rate Interface) unit from here on)converts the signal into a signal having an exchangeable form, andcreates and adds a control signal or the like to the signal. Two or more2.048 MHz (E1) or 1.544 MHz (T1) signal lines which comply with ITU-Trecommendation G.703/G.704 extending from the exchange 10 are inputtedto the PRI unit 1. 30 channels are multiplexed per one voice signal whenthe plurality of signal lines inputted to the PRI unit 1 are E1 signallines, whereas 24 channels are multiplexed per one voice signal when theplurality of signal lines inputted to the PRI unit 1 are T1 signallines. The PRI unit 1 demultiplexes each inputted voice signal into anumber of signals associated with the corresponding number of channels,and outputs them to the voice signal processing unit 2 channel bychannel.

The voice signal processing unit 2 (referred to as the VFP (VoiceFrequency Signal Processing) unit from here on) performs detection ofsound portions or soundless portions, and coding and decoding on eachvoice signal inputted thereto from the PRI unit 1. The VFP unit 2carries out compression coding on only parts of each voice signalinputted thereto from the PRI unit 1, which are determined as soundportions, by means of a high-efficiency voice CODEC (for example, anITU-T recommendation G.729 CS-ACELP CODEC), and outputs each compressedvoice signal to the packet processing unit 3.

The packet processing unit 3 packetizes each voice signal on which thecompression coding has been carried out by the VFP unit 2 into RTP(Real-Time Transfer Protocol), UDP (User Datagram Protocol), and IPpackets which are running in order of RTP, UDP, and IP, and outputs themto the Ethernet I/F processing unit 4 (referred to as the network I/Funit from here on). The packet processing unit 3 also depacketizespackets inputted thereto from the network I/F unit 4 into signals, andoutputs them to the VFP unit 2. The packet processing unit 3 furthercollects information, such as information about basic statistics, andperforms output processing on the collected information. RTP packets aredefined by RFC (Request For Comments) 2508 of IETF (Internet EngineeringTask Force), as shown in FIG. 2.

The network I/F unit 4 converts packets received thereby from the packetprocessing unit 3 into Ethernet frames, and sends out them onto anetwork (for example, an IP network). Thedevice-operation-monitoring-control processing unit (referred to as theMCPU unit (or Main CPU) from here on) 5 downloads various apparatusparameters received thereby from the operation monitor control desk 9 torespective components which constitute the voice transmission apparatus100, and monitors a failure status of each of the components and outputsinformation indicating the failure status to the operation monitorcontrol desk 9. If the voice transmission apparatus 100 has a componentin which a failure has occurred, the device-operation-monitoring-controlprocessing unit carries out operation and monitoring control on thewhole of the apparatus, such as a blockage of a voice output signal lineof the component in which a failure has occurred, so that other normallyoperating units may not be affected. The control processing is performedaccording to three pieces of control information (1), (2), and (3) asshown in FIG. 1.

The amount of discarded voice packets/packet jitter measuring unit 6measures the amount of discarded packets associated with the receivedvoice and the amount of jitter of the received voice based oninformation about the packets processed by the packet processing unit 3.When the amount of packets transmitted via the network is large,discarding of some packets and an amount of jitter may occur in eachrouter or the like arranged on the network.

The calling restriction determining unit 7 determines whether to imposea calling restriction on the subscriber-side exchange 10 based on theamount of discarded packets and the amount of packet jitter which aremeasured by the amount of discarded voice packets/packet jittermeasuring unit 6.

The ITU-T Q.50 DLC processing unit 8 (referred to as the DLC (DynamicLoad Control) control unit from here on) accepts the determinationresult of the calling restriction determining unit 7, and, when thedetermination result indicates that a calling restriction needs to beimposed on the subscriber-side exchange 10, outputs DLC information tothe PRI unit 1 based on ITU-T Q.50 by using a signaling bit to thesubscriber-side exchange 10. The information about this callingrestriction is periodically inputted to the DLC control unit from thecalling restriction determining unit 7.

The operation monitor control desk 9 monitors and controls the operatingstate and operation or use state of the voice transmission apparatus100, and transmits various apparatus parameters to the MCPU unit 5. Thevoice transmission apparatus 100 is controlled according to theseparameters.

The exchange 10 is engaged in exchange of signals between asubscriber-side network and the IP network, and callings are controlledaccording to a calling restriction determination signal from the PRIunit 1.

The above description is directed to a flow of signals from the exchange10, i.e., the subscriber side to the network side. In the case of a flowof signals in an opposite direction, i.e., a direction from the networkside to the subscriber side, the network I/F unit 4 extracts packetsfrom each Ethernet frame inputted thereto from the network side, and thepacket processing unit 3 disassembles the packets from the network I/Funit 4 in order of IP, UDP, and RTP packets so as to extract coded voicesignals. The VFP unit 2 decodes the voice signals extracted by thepacket processing unit 3, and the PRI unit 1 multiplexes the decodedvoice signals transmitted via channels and then outputs them to theexchange 10. The other components operate in the same way as mentionedabove.

Next, a communication system to which one above-mentioned voicetransmission apparatus 100 is applied will be explained. FIG. 3 shows afirst example of the communication system in which subscriber-sideapparatus are connected to an IP network 21 via a plurality of routers22 a to 22 n. A plurality of subscriber-side terminals (for example,telephones and FAX machines) 23 a to 23 n and a plurality of dataterminals 24 a to 24 n are connected to the plurality of routers 22 a to22 n by way of exchanges 25 a to 25 n and media gateways 26 a to 26 n. Aplurality of calling restriction units 27 a to 27 n are disposed in theplurality of media gateways 26 a to 26 n, respectively, and each of theplurality of routers 22 a to 22 n is equipped with a means for measuringthe amount of discarded voice packets and the amount of packet jitter.In this first example, each of the plurality of routers 22 a to 22 n andthe plurality of calling restriction units 27 a to 27 n is provided withthe same components as those included in one voice transmissionapparatus 100 in accordance with the present invention.

Each of the plurality of routers 22 a to 22 n measures the amount ofpackets which are discarded from voice packets which it obtains from theIP network 21 and the amount of packet jitter of the voice packets, andnotifies measurement results to a corresponding one of the plurality ofmedia gateways 26 a to 26 n. Each of the plurality of callingrestriction units 27 a to 27 n provides an instruction for imposing acalling restriction to a corresponding one of the plurality of exchanges25 a to 25 n when either one of the measured values is equal to orgreater than a predetermined threshold value.

Thus, each of the plurality of routers 22 a to 22 n measures the amountof packets which are discarded from voice packets which are included ina large volume of packets transmitted via the IP network 21 fromcorresponding ones of the plurality of subscriber-side terminals 23 a to23 n and the plurality of data terminals 24 a to 24 n, and the amount ofpacket jitter of the voice packets, and imposes a calling restriction ona corresponding one of the plurality of exchanges 25 a to 25 n based onthe measured values. Therefore, the communication system makes itpossible to prevent degradation in the quality of voice signals.

FIG. 4 is a second example of the communication system to which voicetransmission apparatus 100 according to this embodiment are applied, andin which subscriber-side apparatus are connected to the IP network 21via a plurality of routers 22 a to 22 n. A plurality of subscriber-sideterminals (for example, telephones and FAX machines) 23 a to 23 n and aplurality of data terminals 24 a to 24 n are connected to the pluralityof routers 22 a to 22 n by way of exchanges 25 a to 25 n and mediagateways 26 a to 26 n. A plurality of signaling gateways 28 a to 28 nare placed between the plurality of routers 22 a to 22 n and theplurality of exchanges 25 a to 25 n, respectively. Each of the pluralityof routers 22 a to 22 n is equipped with a means for measuring theamount of discarded voice packets and the amount of packet jitter, andeach of the plurality of signaling gateways 28 a to 28 n is providedwith a function of imposing a calling restriction on a corresponding oneof the plurality of exchanges 25 a to 25. In this second example, eachof the plurality of routers 22 a to 22 n and the plurality of signalinggateways 28 a to 28 n is equipped with the same components as those ofone voice transmission apparatus 100 in accordance with the presentinvention.

Each of the plurality of routers 22 a to 22 n measures the amount ofpackets which are discarded from voice packets which it obtains from theIP network 21 and the amount of packet jitter of the voice packets, andnotifies measurement results to a corresponding one of the plurality ofsignaling gateways 28 a to 28 n. Each of the plurality of signalinggateways 28 a to 28 n provides an instruction for imposing a callingrestriction to a corresponding one of the plurality of exchanges 25 a to25 n when either one of the measured values is equal to or greater thana predetermined threshold value.

Thus, each of the plurality of routers 22 a to 22 n measures the amountof packets which are discarded from voice packets included in a largevolume of packets transmitted via the IP network 21 from correspondingones of the plurality of subscriber-side terminals 23 a to 23 n and theplurality of data terminals 24 a to 24 n, and the amount of packetjitter of the voice packets, and each of the plurality of signalinggateways 28 a to 28 n imposes a calling restriction on a correspondingone of the plurality of exchanges 25 a to 25 n based on the measuredvalues. Therefore, the communication system makes it possible to preventdegradation in the quality of voice signals.

Next, the operation of one voice transmission apparatus 100 inaccordance with embodiment 1 will be explained with reference to a flowchart of FIG. 5.

First, the voice transmission apparatus 100 receives voice RTP packetswhich are disassembled by the packet processing unit 3 and which areincluded in data packets obtained from the IP network 21 via the networkI/F unit 4 (in step ST101). The amount of discarded voice packets/packetjitter measuring unit 6 then measures and stores the number of discardedpackets (in step ST102), and also measures and stores the amount ofpacket jitter (in step ST103). The amount of discarded voicepackets/packet jitter measuring unit 6 measures the number of discardedpackets based on sequence numbers, as shown in FIG. 2, included in theheaders of the RTP packets, and measures the amount of packet jitterbased on time stamp information included in the headers of the RTPpackets.

The calling restriction determining unit 7 then determines whether ornot the number of discarded packets is equal to or greater than apredetermined threshold (in step ST104). When the number of discardedpackets is not equal to or greater than the predetermined threshold(i.e., if No), the calling restriction determining unit 7 furtherdetermines whether or not the amount of packet jitter is equal to orgreater than a predetermined threshold (in step ST105). When the callingrestriction determining unit 7, in step ST104, determines that thenumber of discarded packets is equal to or greater than thepredetermined threshold (i.e., if Yes), the DLC control unit 8 creates acalling restriction control signal for imposing a calling restriction onthe exchange 10 (in step ST106). As a result, the PRI unit 1 imposes acalling restriction on the exchange 10 (in step ST107), and the voicetransmission apparatus 100 returns to step ST101 in which it repeats themonitoring process.

When the calling restriction determining unit 7, in step ST105,determines that the amount of packet jitter is equal to or greater thanthe predetermined threshold (i.e., if Yes), the DLC control unit 8 alsocreates a calling restriction control signal for imposing a callingrestriction on the exchange 10 (in step ST106). As a result, the PRIunit 1 imposes a calling restriction on the exchange 10 (in step ST107),and the voice transmission apparatus 100 returns to step ST101 in whichit repeats the monitoring process. In contrast, when determining thatthe amount of packet jitter is less than the predetermined threshold(i.e., if No), the calling restriction determining unit 7 determineswhether or not a calling restriction is being imposed on the exchange(in step ST108). When no calling restriction is being imposed on theexchange (i.e., if No), the voice transmission apparatus 100 returns tostep ST101 in which it repeats the monitoring process. In contrast, whena calling restriction is being imposed on the exchange (i.e., if Yes),the voice transmission apparatus 100 cancels this calling restriction(in step St109) and then returns to step ST101 in which it repeats themonitoring process.

As explained above, even when the components included in the voicetransmission apparatus 100 operate, and some voice packets are discardedand an amount of packet jitter occurs in voice packets, voice data canbe transmitted without causing degradation in the quality of voicesignals.

The flow of the operation of the voice transmission apparatus 100 is notlimited to the above-mentioned one, and can be any one in which similarcontrol processing is carried out.

INDUSTRIAL APPLICABILITY

As mentioned above, the voice transmission apparatus in accordance withthe present invention can be placed between a subscriber-side exchangeto which subscriber-side apparatus are connected and a network, and issuitable for preventing degradation in the quality of voice signalswhich are packetized and are transmitted between the subscriber-sideexchange and the network.

1. A voice transmission apparatus arranged between a subscriber-sideexchange to which a subscriber-side terminal apparatus is connected anda network, characterized in that said voice transmission apparatuscomprises: a first interface means for exchanging a signal with saidsubscriber-side exchange; a second interface means for exchanging asignal with said network; a voice signal processing means for detectingsound portions of a voice signal which constitutes said signal, forencoding the voice signal having sound portions, and for decoding anencoded voice signal inputted thereto; a packet processing means forassembling packets from the voice signal encoded by said voice signalprocessing means, and for disassembling packets inputted thereto into asignal and for supplying it to said voice signal processing means; ameasurement means for measuring an amount of packets which are discardedfrom packets associated with a voice signal included in the receivedpackets, and an amount of jitter of the packets associated with thevoice signal; a calling restriction determination means for determiningwhether to impose a calling restriction based on measurement resultsobtained by said measurement means; and a control signal creating meansfor creating a control signal for imposing a calling restriction on saidsubscriber-side exchange when said calling restriction determinationmeans determines that the calling restriction needs to be imposed onsaid subscriber-side exchange.
 2. The voice transmission apparatusaccording to claim 1, characterized in that the packets include an IPpacket, a UDP packet, and an RTP packet.
 3. The voice transmissionapparatus according to claim 1, characterized in that said secondinterface means is an interface means which complies with Ethernet. 4.The voice transmission apparatus according to claim 1, characterized inthat said control signal output means is an ITU-T Q.50 DLC processingmeans for outputting the control signal for imposing a callingrestriction on said subscriber-side exchange according to ITU-Trecommendation Q.50.